Method for the integrated transmission of first data with real-time requirement and second data without real-time requirement, communication device and communications system

ABSTRACT

A plurality of first quality of service classes is provided for transmitting first data and a plurality of second quality of service classes is provided for transmitting second data, in each case in the application layer. A combined quality of service class is selected from the combined quality of service classes formed from the first quality of service classes and the second quality of service classes. The first data and the second data are supplied to a unit of the transport layer, and the unit transmits the data in dependence on the transmission parameters allocated to the selected combined quality of service class.

BACKGROUND OF THE INVENTION

[0001] Field of the Invention

[0002] The invention relates to a method for the integrated transmissionof first data with real-time requirement and second data withoutreal-time requirement, a communication device and a communicationssystem.

[0003] The Open System Interconnection (OSI) layer model defined by theInternational Organization for Standardization (ISO) is described by A.S. Tanenbaum, Computer-Netzwerke [Computer Networks], Wolframs'sFachverlag, 2^(nd) Ed., ISBN 3-925328-79-3, p. 17-32, 1992. Within theOSI layer model, different tasks existing in a transmission of databetween computers, generally a communication between computers in aheterogeneous communication network, are distributed over differentlayers which in each case provide predetermined services transparently,that is to say they are provided by a unit in the respective layer to aunit of a “layer above” in such a manner that the layers above cannotsee how the respective service is provided but only that the respectiveservice is provided.

[0004] In the OSI layer model, in particular, a distinction is madebetween layers which are responsible for the error-free transmission ofdata from one computer to another computer within the communicationnetwork and layers which use these services.

[0005] A layer which has the task of ensuring end-to-end communicationfrom a transmitting computer to a receiving computer, is the so-calledtransport layer (layer 4 in the OSI layer model).

[0006] An example of a protocol of the transport layer is the TransportControl Protocol (TCP) which usually operates in conjunction with theInternet Protocol (IP) of the network layer (layer 3 of the OSI layermodel). It is also known to provide in the IP a transmission of the datain accordance with different quality of service classes (COF forintegrated services) at the level of the network layer.

[0007] A distinction must be made between the transport layer and, forexample, the application layer (layer 7 in the OSI layer model), inwhich the transmission of data by means of a communication protocol isdetermined purely from the point of view of application and not from apoint of view in which the individual transmission components are takeninto consideration. To illustrate, the application layer usuallycontains the user program.

[0008] Examples of protocols in the application layer are the HypertextTransfer Protocol (HTTP), the method according to a MPEG standard forcoding video data and a method for transmitting still pictures, forexample the method according to the JPEG standard. Furthermore, methodsfor coding voice data in accordance with the TIPHON standard (see ETSITIPHON, Telecommunications and Internet Protocol Harmonization OverNetworks, General Aspects of Quality of Service (QoS), TR 101 329 V2.1.1(1999-06), June 1999) are defined in the application layer.

[0009] In the case of multimedia data to be transmitted, a distinctionmust be made between data to be transmitted in a case of which, inparticular, a very high demand must be made that it can be guaranteed,that the delay time between the data is very short as is the case, forexample, with voice data or video data. In the case of voice data, it isparticularly important that the voice data transmitted are received atthe receiving computer with very little delay since otherwise thequality of the received reconstructed voice data is considerably reducedfor the user of the receiving computer who listens to the reconstructedvoice data. In particular, such requirements will also be calledreal-time requirements of the data in the text which follows.

[0010] In contrast, a usual multimedia data stream also contains seconddata without real-time requirements, for example text data or alsostill-image data.

[0011] In the case of such data, it is only generally important that thedata are transmitted as free of errors as possible but not necessarilythat, for example, the delay of the transmission between the individualdata elements is as short as possible.

[0012] It has been known to provide in each case a communication linkfor first data with real-time requirement and second data withoutreal-time requirement. See, for example, IETF working group, PSTN andInternet Internetworking (pint), available on April 2, 2000 at the URLaddress http://www.ietf.org/html.charters/pint-charter.html, and WAP:Wireless Telephony Application Specification, available on Apr. 2, 2000at URL addresshttp://ww1.wapforum.org/tech/documents/SPEC-WTA-19991108.pdf.

[0013] It must also be noted that, in the case of mobile communicationterminals, it is usually often impossible or at least impractical to setup separate communication links for first data with real-timerequirement and second data without real-time requirement over a numberof mobile terminals as is provided in accordance with the prior artdescribed in the above IETF working group publication. In the case ofmobile communication terminals, in particular, this is attributable tothe scarcity of resources of the mobile communication terminals and ofthe available bandwidth in a mobile radio network which is much lessthan in the case of a communications system in which only a landlinenetwork is provided.

[0014] To achieve further integration of the transmission of multimediadata, it is known from Krautgärtner and Decker, et al., in “Design ofV/D-API and Architecture of the VE-MASE”, CEC Deliverable NumberAC343/Siemens/WP2/DS/P/02/a1, Project Number AC343, November 1998, totransmit first data with real-time requirement and second data withoutreal-time requirement in an integrated manner from the point of view ofthe application layer in a communication link.

[0015] In the prior art communications system according to the MOVEarchitecture, the middleware VE-MASE (Voice Enabled Mobile ApplicationSupport Environment), as defined by Krautgärtner and Decker, which isinstalled in client computers which are located in a predeterminedcommunication network, and in gateways which are used as switchingcomputers between a wire-connected part and a wireless part of acommunication network, and in mobile communication terminals such as,for example, a notebook, a Personal Digital Assistant (PDA) or also amobile radio telephone with which communication is possible inaccordance with the Wireless Access Protocol (WAP).

[0016] It is also known from the above ETSI TIPHON paper to divide firstdata with real-time requirements into different quality of service (QoS)classes. Each quality of service class of the first data is in each caseallocated to a predetermined guaranteed quality of the reconstructedvoice data, for example the delay to be guaranteed, the time for theconnection set-up and other mechanisms for guaranteeing a predeterminedquality of the voice signal.

[0017] Bhatti and Knight, in “Enabling QoS Adaptations for InternetApplications”, Journal of Computer Networks, Vol. 31, No. 7, p. 669-92,March 1999, describe mapping the transmission of data from theapplication layer to the transport layer for only one type of data,namely for the first data with real-time requirement—voice data in themethod described bz Bhatti and Knight. In particular, the choice ofvoice codec (coder and decoder) to be used, as a function of measuredtransmission parameters, that is to say the throughput, the jitter andthe delay of the data to be transmitted, is described.

[0018] Furthermore, a general classification scheme at nontechnicallevel for first data with real-time requirement and second data withoutreal-time requirement is known from Decker, Krautgärtner, Ong, andWallbaum, in “Quality of Service Management in an Integrated MobileVoice/Data-Enabled Service Architecture,” Proceedings of the 4^(th) ACTSMobile Communication Summit, Jun. 8-11, 1999, Sorrento, Italy; and fromDecker and Krautgärtner, “Flexible Quality-of-Service Technology forSupporting Voice/Data-Integrated Nomadic Networking,” inChistensen-Dalsgaard, Donelly, and Griffith (eds), Flexible Working—NewNetwork Technologies, IOS Press/Ohmsha, 1999, ISBN 1 58603 028 0 (IOSPress), ISBN 4 274 90322 2 C3050 (Ohmsha), Library of Congress CardNumber 99-67675.

[0019] The so-called Session Initiation Protocol (SIP) is known fromHandley, et al., SIP: Session Initiation Protocol, IETF Request forComments 2543, March 1999.

SUMMARY OF THE INVENTION

[0020] The invention is based on the object of providing a method oftransmitting first data with real-time requirement and second datawithout real-time requirement in a simple manner, taking intoconsideration their requirement for the transmission quality, in theapplication layer in, from the point of view of the application layer, asingle communication link. It is a further object to provide acorresponding communications system and a communications device.

[0021] With the above and other objects in view there is provided, inaccordance with the invention, a method of transmitting data withreal-time requirement and data without real-time requirement, whichcomprises:

[0022] providing a plurality of first quality of service classes in anapplication layer for transmitting first data with real-time requirementand a plurality of second quality of service classes in the applicationlayer for transmitting second data without real-time requirement;

[0023] selecting a combined quality of service class formed from thefirst quality of service classes and the second quality of serviceclasses in the application layer, each combined quality of service classbeing allocated transmission parameters specifying a transmission of thefirst data and the second data; and

[0024] supplying the first data and the second data and the transmissionparameters of the selected combined quality of service class to a unitof a transport layer, and transmitting the first data and the seconddata with the unit taking into consideration the transmissionparameters.

[0025] In other words, in the method for transmitting first data withreal-time requirement and second data without real-time requirement, aplurality of first quality of service classes is allocated to the firstdata at the level of the application layer for transmitting the firstdata. For transmitting the second data, a plurality of second quality ofservice classes is allocated to the second data at the level of theapplication layer. One of the combined quality of service classes formedfrom the first quality of service classes and the second quality ofservice classes is selected in the application layer, each combinedquality of service class being allocated transmission parameters bymeans of which it is specified by means of which parameters the firstdata and the second data are to be transmitted. In particular, aprioritization of the transmission of the first data relative to thesecond data and conversely is provided in the transmission parameters ofthe respective combined quality of service classes. The first data andthe second data and the transmission parameters of the selected combinedquality of service class are supplied to a unit of the transport layerby means of which the first data and the second data are transmittedtaking into consideration the transmission parameters.

[0026] The first data with real-time requirement are, for example, voicedata in which the first quality of service classes according to TIPHONas described in the above-quoted ETSI TIPHON paper can be allocated. Inaddition, another additional quality of service class can be providedfor further differentiation for improved adaptation of the quality ofservice classes from TIPHON to the requirements of the integratedtransmission of data with real-time requirement and data withoutreal-time requirement.

[0027] Furthermore, the first data can also contain video data,particularly data in accordance with the MPEG standard or a videotelephone standard such as, for example, in accordance with theITU-H.263 standard.

[0028] The second data without real-time requirement can be, forexample, text data which are transmitted, for example, in accordancewith the HTTP protocol or in accordance with the WAP protocol, or alsostill-image data, for example according to the JPEG standard.

[0029] The second quality of service classes can be predetermined in asimilar manner to the first quality of service classes in accordancewith the requirements set for the transmission of the first data and theresources of the transmission medium needed for guaranteeing therequirements.

[0030] From the first quality of service classes and the second qualityof service classes, combined quality of service classes are formed fromwhich a combined quality of service class is selected and thetransmission parameters allocated to the respective selected combinedquality of service class are supplied to the unit of the transportlayer. The respective quality of service classes can be allocated ineach case a priority which specifies the priority with which therespective data are to be transmitted. The combined quality of serviceclasses can be formed as a function of the first priorities and thesecond priorities which are allocated to the first quality of serviceclasses or, respectively, to the second quality of service classes.

[0031] The combined quality of service class can be selected in thefollowing manner:

[0032] a) the combined quality of service class which has the firstquality of service class with the highest first priority and the secondquality of service class with the highest second priority is selected,

[0033] b) a check is made whether a coder to be used can transmit thefirst data and the second data according to the transmission parametersof the respective combined quality of service class,

[0034] c) if this is so, the combined quality of service class isselected,

[0035] d) if this is not so, a further combined quality of service classis selected in such a manner that in each case the combined quality ofservice class with reduced second priority is selected, and

[0036] e) steps b) and d) are performed iteratively until the coder cantransmit the first data and the second data in accordance with thetransmission parameters of the respective combined quality of serviceclass.

[0037] In this manner, very simple heuristics are specified according towhich the combined quality of service class can be selected. Thesesimple heuristics are distinguished by their low requirement forcomputing time, especially in the case of mobile communicationterminals, for example a mobile telephone, in which only relatively lowcomputing capacity is available.

[0038] By comparison, heuristics based on statistical models knownfrequently from different optimization problems are much morecomputer-intensive and can scarcely be used in practice for acommunication terminal having low available computing capacity.

[0039] In the transport layer unit, the first and the second data can becoded and transmitted as a data stream with predeterminable transportlayer quality of service class.

[0040] A communication device, preferably a mobile communication device,has a processor which is set up in such a manner that the method stepsdescribed above can be performed.

[0041] Owing to its simplicity and the low requirements for thecomputing capacity needed, the method described above is very suitable,in particular, for a mobile communication device.

[0042] In a communications system, a mobile first communication deviceand a second communication device is provided. The first data and thesecond data are transmitted from the first communication device to thesecond communication device.

[0043] It must be noted that, in particular, voice data are verysensitive also to mobile-radio-related fluctuations in the bandwidth andthe delay or the variance of the voice data to be transmitted.

[0044] A considerable advantage of the invention can be seen in the factthat, particularly in heterogeneous mobile radio networks, that is tosay in mobile radio networks having different mobile radio subnetworks,for example an inexpensive proprietary mobile radio network of auniversity having a large available bandwidth and a public UMTS mobileradio network having a relatively little available bandwidth, anadaptation becomes necessary either of the end-to-end quality of thetransmitted data directly in the communication terminal or duringrecoding from one voice/video codec to another one or during filteringof the data in the network on transmission from one mobile radio networkto the other, which is also called handover.

[0045] The invention now makes it possible to adapt and to vary thetransmission of the data at application protocol level on the basis ofclassifications which are predetermined for a user and are generallyunderstandably, including currently measured QoS parameters such as, forexample, the delay, the jitter and the available bandwidth, and to mapit to real parameters describing the respective data stream and thetransmission parameters, such as, for example, the parameters whichdescribe the codec to be used for coding, the transmission formats usedand/or the compression parameters for video data, image data or textdata and to vary these as a function of application.

[0046] The invention is particularly suitable for use in internettelephony, for example in the field of so-called voice over IP (VOIP) orvideo over IP via communication links for transmitting data which areoriginally text data. One such possibility is, for example, providedexclusively as multimedia domain in mobile radio standard 3GPP Release00 (UMTS).

[0047] Furthermore, a multiplicity of different voice codecs, generallycodecs for data with real-time requirement and/or without real-timerequirement can be stored either permanently in the communicationterminals or can also be loaded temporarily into the communicationterminal.

[0048] The invention also makes it possible to perform a quite specificprioritization of different information data streams and theiradaptation depending on the requirements of a given application. Thus,it may be possible in an application that the data with real-timerequirement are more important than data without real-time requirement.However, the case may also occur where the data without real-timerequirement has to be given a higher priority in the data transmissionthan data with real-time requirement.

[0049] The invention makes it possible to specify this even at the levelof the application layer.

[0050] According to the invention, it is also possible to respondquickly and flexibly to changing transmission conditions of the codecused or the communication network used, respectively, at applicationprotocol level.

[0051] Thus, the quality of service of the combined voice services/dataservices can already be adapted a priori if there is knowledge about theavailable bandwidth being too little, for example during handover.

[0052] This leads to a further improvement in the total performance ofthe available communications system.

[0053] Other features which are considered as characteristic for theinvention are set forth in the appended claims.

[0054] Although the invention is illustrated and described herein asembodied in a method for the integrated transmission of first data withreal-time requirement and second data without real-time requirement,communication device and communications system, it is nevertheless notintended to be limited to the details shown, since various modificationsand structural changes may be made therein without departing from thespirit of the invention and within the scope and range of equivalents ofthe claims.

[0055] The construction and method of operation of the invention,however, together with additional objects and advantages thereof will bebest understood from the following description of specific embodimentswhen read in connection with the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

[0056]FIG. 1 is a flowchart in which the individual method stepsaccording to the invention are shown in accordance with an exemplaryembodiment of the invention;

[0057]FIG. 2 is a block diagram showing the individual elements of acommunications system according to the MOVE architecture as provided inthe communications system according to the exemplary embodiment of theinvention;

[0058]FIG. 3 is a block diagram illustrating the individual componentsof the VE-MASE middleware according to the MOVE architecture within acommunication device as used in accordance with an exemplary embodimentof the invention; and

[0059]FIG. 4 is a table listing the combined quality of service classesaccording to an exemplary embodiment of the invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0060] Referring now to the figures of the drawing in detail and first,particularly, to FIG. 2 thereof, there is seen a communications system200 comprising a mobile communication device 201, a notebook, a PDA, ora WAP-capable mobile radio telephone according to the exemplaryembodiment.

[0061] As is shown symbolically in FIG. 2, the mobile communciationdevice 201 has an input/output interface 202 and a communication callmanaging unit 203 (called call manager in the MOVE architecture).

[0062] Furthermore, a Voice over IP client 204 is installed in themobile communication device 201.

[0063] Furthermore, a browser program 205 which can represent and codedata according to the HTML format or the WML format is installed in themobile communication device 201. In the Voice over IP client 204, thedata are coded in accordance with the UDP/IP protocols in the transportlayer or in the network layer according to the OSI layer model.

[0064] It is noted, in this context, that the Voice over IP client andthe browser program act independently of one another. The mobilecommunication device 201 is coupled to a switching computer 220 via aradio link 210, a radio link according to the UMTS standard according tothe exemplary embodiment.

[0065] The radio link can be, for example, a connection in a wirelessLocal Area Network (wireless LAN), a connection according to the DECT,the GSM, or the UMTS.

[0066] The switching computer 220 also has an input/output interface221, a call manager 222 and a communication link adaptation unit(collaboration manager) 223, an audio interface 224, a HTTP proxy unitand a scheduler 226.

[0067] Furthermore, a further computer is coupled as secondcommunication device 240 to the switching computer 220 via a landlinelink 230.

[0068] A communication link is set up between the mobile communicationdevice 201 and the second communication device 240 via the radio link210, the switching computer 220 and the landline link 230 andcommunication takes place via the communication link setup.

[0069] The second communication device 240 also has an input/outputinterface 241 and a call manager 242. Furthermore, a Voice over IPclient 243 and a browser program 244, which can communicate with thecorresponding unit of the mobile communication device 201 in accordancewith a predetermined protocol in the application layer or, respectively,the transport layer is installed in the second communication device 240.

[0070] The VE-MASE middleware of the MOVE architecture, described in thetext which follows and shown in FIG. 3, is installed both in thecommunication devices 201, 240 and in the switching computer 220.

[0071] As will be described in the following text, the middlewareensures the following services:

[0072] the transmission of data from and to the respective communicationdevice in accordance with a HTTP protocol or the WAP protocol; and

[0073] the transmission of audio data via a mobile radio network,guaranteeing both the recoding and the scheduling of real-time data andnon-real-time data, that is to say of first data with real-timerequirement and of second data without real-time requirement.

[0074] In the text which follows, some components of the devicesaccording to the MOVE architecture described and defined in detail inthe above-mentioned article by Krautgärtner and Decker, et al., Designof V/D-API and Architecture of the VE-MASE, CEC Deliverable NumberAC343, will be explained in an overview.

[0075] The audio gateway provides the service of a real-time audioconference between units which communicate via a mobile radio link andunits which communicate with one another in a landline network and unitswhich are located within a landline network and communicate via alandline network link.

[0076] The audio gateway adapts first data, that is to say audio data inthis case, especially voice data, in accordance with changingrequirements by means of changed codecs and their parameters, by noisesuppression and by recoding the respective data streams.

[0077] The scheduler according to the exemplary embodiment of theinvention ensures that the first data with real-time requirement, thatis to say the voice data and the video data, are not delayed by seconddata without real-time requirement, for example text data such as dataaccording to the HTML format or electronic mail, audio data usuallybeing assigned higher priority than video data in the transmission.

[0078] Data packets arriving at the scheduler are divided in accordancewith their quality of service class allocated to the respective datapacket.

[0079] Furthermore, the scheduler measures quality of service parametersfor each data stream and the results of the measurement are supplied tothe collaboration manager 223.

[0080] The call manager is distributed over the communication devicesconnected within the communications system 2000.

[0081] The call manager is responsible for

[0082] setting up a communication link;

[0083] controlling the communication link; and

[0084] ending the communication link in which both first data withreal-time requirement and second data without real-time requirement areexchanged between the mobile communication device 201 and the secondcommunication device 240.

[0085] The call manager 203 is set up in such a manner that the SessionInitiation Protocol SIP is implemented as described in the above-quotedM. Handley et al., SIP: Session Initiation Protocol, IETF Request forComments 2543, March 1999.

[0086] The HTTP proxy maps data according to the HTML format to therequirements or capabilities corresponding to the respectivecommunication device to which the data are to be sent. Thus, the HTTPproxy 225 suppresses color information, if necessary, that is to saycolor images are mapped onto black/white images if the receiver devicecan only display black/white images. Furthermore, different scalings andresolutions of the information can be changed or even complete videodata or image data are filtered out, for example on transition from anHTML data format to a WML data format.

[0087] Further details can be found in the description of the MOVEarchitecture as defined in Krautgärtner and Decker, et al., Design ofV/D-API and Architecture of the VE-MASE, CEC Deliverable Number AC343.

[0088]FIG. 3 shows an overview of the individual elements and theirsymbolic arrangement within the layer model for a communication device.

[0089] The individual application programs 302, for example theworldwide web browser, various call center services or also, forexample, a program for performing Internet radio, are stored in theapplication layer 301 of the communication device 300.

[0090] A voice/data interface 303 and a mobile radio interface 304 areprovided in accordance with the VE-MASE architecture as described inKrautgärtner and Decker, et al., Design of V/D-API and Architecture ofthe VE-MASE, CEC Deliverable Number AC343.

[0091] The following components are provided in the application layer301

[0092] the call manager 305 for setting up and managing a communicationlink, a collaboration manager 306 for managing the web data streams,

[0093] the session adaptation manager 307, and

[0094] the audio gateway 308.

[0095] The following components are provided in the mobile radiointerface 304:

[0096] a profile manager 309 for managing the profiles of the possiblecommunication partners,

[0097] a location manager 310,

[0098] a multimedia conversion unit 311,

[0099] a user manager 312,

[0100] a directory services unit 313,

[0101] an event manager 314.

[0102] The units of the application layer 301 are coupled to units of aUMTS application layer 320.

[0103] The units of the UMTS adaptation layer 320 are coupled todifferent mobile radio networks 330, the units of the UMTS adaptationlayer being set up in such a manner that the data of the applicationlayer 301 are appropriately mapped to the required formats which arerequired for transmission depending on the (mobile) radio data formatused.

[0104] The (mobile) radio protocol used can be, for example, theprotocol according to GSM standard 331, or also the protocol accordingto the wireless transmission protocol DECT 332 or the protocol accordingto the wireless LAN 333 or also the protocol according to IMT 2000 UMTS334.

[0105] In the text which follows, an overview of the functionality ofthe session adaptation manager 307 is given.

[0106] The session adaptation manager 307 selects one of variousavailable communication networks, that is to say various wirelesscommunication links used according to different transmission protocols,in such a manner that the selected communication network meetspredetermined quality of service requirements, for example an availablebandwidth or also a predetermined price of a communication link, that isto say a predetermined tariff.

[0107] The session adaptation manager 307 can generate signals by meansof which the manner is specified in which data to be transmitted arecompressed, converted, transcoded or even individual multimedia objects,for example individual images or videos, are filtered out beforetransmission.

[0108] If, for example, image data or video data are intended to betransmitted to a communication device which can display only image dataor video data in black/white, data representing color information areremoved in accordance with the control signals of the session adaptationmanager 307 as a result of which the bandwidth needed is reduced.

[0109] In general, the session adaptation manager 307 selects, as willbe explained in greater detail in the text which follows, a combinedquality of service class which establishes transmission parameters bymeans of which the data to be transmitted are transmitted taking intoconsideration the boundary conditions determined by the transmissionparameters.

[0110] Thus, for example if the available bandwidth is reduced in awireless communication link, for example due to changed selection of acodex for coding audio data, by requesting a reduction in the magnitudeof HTML data at the multimedia conversion proxy, by suppressing voicetransmission, by suppressing the possibility of exchanging data inaccordance with the HTTP protocol or the WAP protocol, an adaptation ofthe available transmission resources in the application layer to therequirements of the data to be transmitted is achieved.

[0111] Furthermore, it is assumed that first data with real-timerequirement, voice data according to the exemplary embodiment, andsecond data without real-time requirement, text data according to theexemplary embodiment which are coded, for example, in accordance withthe ASCII format, are to be transmitted by the mobile communicationdevice 300.

[0112] The following five quality of service classes are allocated tothe voice data, that is to say to the first data:

[0113] I. Quality of Service Class 5 (least acceptable quality):

[0114] according to the first quality of service class, a communicationlink is guaranteed which is provided in accordance with the availabletransmission resources. According to the first quality of service class,delays of the voice signal can occur at the receiver during thetransmission of the voice data. With a high network load, delays canoccur which lead to the quality of the transmitted voice data beingbelow a usual quality of transmitted voice data in a cellular mobileradio network which is usually guaranteed.

[0115] II. Quality of Service Class 4 (low quality):

[0116] according to a second quality of service class, a somewhatimproved quality of the transmitted voice data is guaranteed.

[0117] III. Quality of Service Class 3 (medium quality):

[0118] a medium quality of the transmitted voice data is guaranteedaccording to a third quality of service class, the medium qualityessentially corresponding to the quality of a usual wireless mobileradio network. The quality is guaranteed, for example, by means of acommunication link via a usual IP communication link.

[0119] IV. Quality of Service Class 2 (high quality):

[0120] according to a fourth quality of service class, a quality of thevoice data to be transmitted is guaranteed which corresponds to that ofa usual landline network telephone connection having a slightly delayed,that is to say increased delay of the voice data.

[0121] V. Quality of Service Class 1 (maximum quality):

[0122] a fifth quality of service class demands the transmission of thevoice data in maximum quality which can be guaranteed at all accordingto the communication network to be used. The quality is at least as goodas the quality of a usual landline network telephone link.

[0123] As described in the above-quoted ETSI TIPHON reference, thequality of the voice data can be categorized by means of a so-called“Mean Opinion Score” (MOS) which reproduces a subjective sensation ofquality by a multiplicity of test persons of the voice data represented.

[0124] The quality scale of the MOS is between a value 1, whichdescribes an unacceptable quality of the voice signal, and a value 5,which represents an excellent quality of voice data.

[0125] A usual telephone codec achieves an MOS of approximately 4.0.

[0126] The first quality of service classes described above are shown inthe table below in an overview with the MOS to be achieved in each case.

[0127] The table also specifies for each first quality of service classthe mean delay time which must be guaranteed, calculated from the timeat which the speaker is speaking the voice signal and the time at whichthe transmitted reconstructed voice signal is output by the receiverdevice, called mouth-to-ear delay in the table.

[0128] Furthermore, a maximum period is specified which is allowed to beneeded for setting up the connection of the communication link, calledcall set-up in the table. Least acceptable Maximum 1 High 2 Medium 3 Low4 5 MOS 4.2-5.0 3.4-4.2 2.6-3.4 1.8-2.6 1.0-1.8 quality Mouth-to- 0-150ms 150-250 250-350 350-500 ≧500 ms ear delay Call set-up 0-1 sec 1-3 sec3-5 sec 5-10 sec ≧10 sec

[0129] The quality of service classes allocated to the first data willbe called first quality of service classes in the text which follows.

[0130] The second data without real-time requirement, that is to say thetext data, are also allocated quality of service classes which will becalled second quality of service classes in the text which follows.

[0131] The second quality of service classes differ with respect to theerror probability to be guaranteed which is allowed to occur at amaximum in a communication link.

[0132] The text which follows is based on the assumption of four secondquality of service classes.

[0133] The case that second data can and are to be transmitted (CWB/MMC)can be refined further into the following conversion forms for imagedata:

[0134] i) there is no color conversion (color quality of the transmittedimage information of the transmitter arrives unchanged at the receiver);

[0135] ii) polychromatic image to four colors (a wide color spectrum ismapped onto a coarse raster);

[0136] iii) colors, i.e. color information is mapped onto gray scalesteps;

[0137] iv) colors or gray scale values of the color information aremapped onto black/white information (such as, e.g. when mapping HTMLimages onto images which are shown according to the WAP standard on aWAP mobile).

[0138] Bandwidth fluctuations can be compensated for by refineddegradations in the spectrum of possible quality steps in the colorconversion (generally the multimedia conversion).

[0139] Orthogonally to the color, the graininess (image resolution) orthe image size, for example, can also be scaled. Scalings in the senseof magnifications or reductions in size are possible and are implementedby means of various MMC (Multimedia Conversion) modes.

[0140] In this case, too, the following applies: the finer theresolution or the larger the transmitted image, the more bandwidth isrequired.

[0141] Thus, in this case, too, a transmission capacity fluctuation canbe compensated for by adapting the quality of multimedia conversion.

[0142] In summary, it can be said that classes 1.x, 3 and 5 in FIG. 4can be refined in accordance with various multimedia dimensions such aschromaticity, granularity, size etc., in any combinations.

[0143]FIG. 4 shows a table 500 in which individual combined quality ofservice classes are shown which are the result of combining theindividual first quality of service classes and second quality ofservice classes.

[0144] As can be seen in FIG. 4, a distinction can be made in threecategories with respect to the voice data according to the exemplaryembodiment.

[0145] In a first category, it is assumed that the respective secondquality of service class can be guaranteed during a communication link,the predetermined delay time of the received voice data never beingexceeded.

[0146] In a second category, it is only possible to guarantee the firstquality of service class of lowest quality.

[0147] In a third category, it is specified that no voice data can betransmitted via the communications network.

[0148] The first category is identified by numbers 1 to 4 in the “oice”column in FIG. 4, the second case is identified by the number 5 and thethird case is identified by the expression “no voice data.”

[0149] Furthermore, a distinction is made in combined quality of serviceclasses K whether text data, that is to say second data withoutreal-time requirements, also need to be transmitted.

[0150] This distinction is shown in the “text data” column of FIG. 4.

[0151] If second data without real-time requirement are to betransmitted, this is marked by “CWB/MMC” (Collaborative WebBrowsing/Multimedia Conversion) in the “text data” column of table 400.

[0152] This provides a statement of the purpose for which non-real-timedata, i.e. second data, are typically transmitted at all and how theyare transmitted, namely, for example, for the purpose of joint(collaborative) viewing and processing of image data, with the aid ofconversion techniques for multimedia data (for instance conversion fromcolor to gray scale representation or masking out of sounds or movingapplets in HTML documents if necessary).

[0153] If no second data are to be transmitted in a communication link,this is specified by the expression “no text data” in table 400.

[0154] If first data and/or second data are transmitted by thecommunication device, the session adaptation manager 307 selects acombined quality of service class K of the combined quality of serviceclasses K shown in FIG. 4.

[0155] The selection is made cyclically at a time interval of someseconds by the session adaptation manager 307.

[0156] In each cycle described in the text which follows, a codecoptimized with regard to the combined quality of service class K and theavailable bandwidth is determined for coding the first data and thesecond data which uses the currently available bandwidth as well aspossible and provides for an optimized transmission rate.

[0157] The selection is made as a function of different quality ofservice parameters which, according to the exemplary embodiment, can beessentially divided into three classes from which the optimized codec isdetermined:

[0158] A first class of quality of service parameters are profileparameters, that is to say quality of service parameters, those of theusers of the communication terminals, the communication terminalsthemselves or characteristic data determined by or dependent on therespective application. Examples of such profile parameters arepriorities for the relative preferential treatment for the first datawith real-time requirement or second data without real-time requirement,information on the discrimination against or equal treatment of firstdata with real-time requirement and second data without real-timerequirement, information on a possible necessity for convertingmultimedia data, for example information on the reduction of HiFiquality of audio data for lower-quality loudspeakers or headphones, theconversion of color information to brightness information, that is tosay to gray scale values for black/white screens, information on maximumtolerable delay values of the received data, that is to say of the datato be transmitted, etc. Profile parameters can also contain otherboundary conditions, for example user-definable parameters,application-dependent parameters or defaults which are caused by thecommunication terminal itself such as, for example, negotiated tariffswhich are tied to certain maximum guaranteed quality of service classes.

[0159] Other quality of service parameters can be parameters which arestored for the duration of one cycle, have been reinitialized in a newcycle, or quality of service parameter values which have been valid forsome preceding cycles. Examples of such quality of service parametersare, in particular, the stored type of the communication network whichis used in the communication link, for example GSM, DECT, HSCSD,wireless LAN etc., and the values of the codec used, which weredetermined in the preceding cycle in time, of the transmission rates andloss rates of data packets in the communication link.

[0160] Other quality of service parameters can be parameters of acommunication link measured by the audio gateway, for example currentvalues of available currently determined bandwidths in communicationlinks to local and remote network nodes, that is to say switchingcomputers, and also local and remote jitter values and delay values inthe transmission of data by the corresponding communication link, and anaverage length of information stream backlog, etc.

[0161] In the text which follows, the method according to the exemplaryembodiment of the invention for selecting a combined quality of serviceclass K is explained in detail with reference to the flowchart of FIG.1.

[0162] The cycle is started in a first step (step 100).

[0163] The type of data to be transmitted is determined in a furtherstep (step 101).

[0164] Furthermore, it is assumed that first data with real-timerequirement and second data without real-time requirement are to betransmitted in the communication link.

[0165] In a further step (step 102), the current profile parameters ofthe mobile communication device, among them the desired prioritiesallocated to the first data or the second data, respectively, aredetermined for an unambiguously identifiable communication link in whichtwo or a multiplicity of communication terminals can communicate withone another, of which at least one communication device is a mobilecommunication device.

[0166] In a further step (step 103), the quality of service parametersof the second class shown above are read in, among these

[0167] the type of communication network used in the communication link,previously stored quality of service features,

[0168] the local codec previously used in the preceding cycle, which isdescribed by means of coding parameters,

[0169] the codec of the remote communication partner, that is to say ofthe respective remote communication device, used in the preceding cycle,

[0170] the stored local loss rate, and

[0171] the stored loss rate of the remote communication partner in thecontext of the preceding cycle.

[0172] In a further step (step 104), the quality of service parametersin class three shown above, provided, for example, by the audio gateway,are determined, that is to say, for example, the type of multimediaconversion if this is dynamic, the available delay values, the thresholdvalues etc.

[0173] In a further step (step 105), a quality of service variabletryQoS is initialized with a maximum value.

[0174] The quality of service variable tryQoS is a value which is shownin the left-hand column in FIG. 4 for identifying the combined qualityof service class K.

[0175] According to the heuristics used in accordance with the exemplaryembodiment, the quality of service variable tryQoS is initialized, on atrial bases, with as high a value as possible as a function of thecurrent profile parameters read in.

[0176] As will be described in the text which follows, the value of thequality of service variable tryQoS is gradually reduced if, according toall other current quality of service parameters, there is no suitablecodec which guarantees the requirements of the current combined qualityof service class K, until a suitable codec has been determined or untilall available codecs have been unsuccessfully tried.

[0177] Naturally, it is possible that there can be a number of suitablecodecs for each combined quality of service class depending on whetherthe corresponding codecs are implemented in the communication device.

[0178] As long as no suitable codec has yet been determined (while loop106), the following method is performed:

[0179] In a first step, a check is made whether the quality of servicevariable is ≧1.1 and is ≦4. This corresponds to the case that atransmission of voice data, generally of first data with real-timerequirement, is possible at all.

[0180] In the text which follows, it is determined, for all availablecodecs i which perform a transmission according to the combined qualityof service class designated by the quality of service variable tryQoS,whether the respective codec i can provide the requested services.

[0181] If this is so, the method finds suitable codecs. The codecs,together with the frames per IP data packet belonging to the codec i,are stored in a list and a check is made whether the bandwidth requiredby the respective codec i is available depending on the availablebandwidth. After that, a list of possible data parameter sets (e.g.according to the quality of service class for image data) is determinedin accordance with the resultant bandwidth for data and personalpreferences of the user.

[0182] After that, the respective optimum combination is selected fromthe two lists (codec, image data parameters) in a further step.

[0183] In the case of the combined quality of service classes K 1.X(high voice data priority relative to the image data), the codec ihaving the highest voice quality (described by the parameter MOSaccording to the present exemplary embodiment) and the lowest number offrames per IP data packet which is compatible with the bandwidth isselected.

[0184] For the image data, a conversion arrangement is selected as afunction of the resultant bandwidths. If this is very narrow, the imagesare only converted in accordance with class iv) of colors or gray scalevalues to black/white image information.

[0185] In the case of the combined quality of service class K 3 (lowvoice data priority relative to the image data), the codec i having thelowest voice quality and the highest number of frames per IP data packetwhich is compatible with the bandwidth is selected. The image data aretransmitted in the best possible quality.

[0186] For quality of service class K 2.x and 4 (no image data), onlythe codec and its parameters are optimized.

[0187] If it is not possible to determine for the respective quality ofservice variable a codec which can guarantee the combined quality ofservice class, i.e. the corresponding requirements, taking intoconsideration the available bandwidth, the value of the quality ofservice variable tryQoS is reduced in accordance with the followingorder of combined quality of service classes to be investigated:

1.1→1.2→1.3→1.4→3→5

[0188] or, respectively,

2.1→2.2→2.3→2.4→4→6.

[0189] These heuristics, which determine the order of the combinedquality of service classes to be investigated, graphically mean that acheck whether there is sufficient bandwidth available for the respectivecodec is made in each case for a combined quality of service classbeginning with the best combined quality of service class having themaximum requirement for bandwidth.

[0190] If this is not so, a combined quality of service class which isin each case worse by one class is selected and a check is made whethera codec is available for this combined quality of service class andwhether sufficient bandwidth can be provided for this codec by thecommunication network.

[0191] This procedure is performed until a codec is determined whichguarantees both the respective current combined quality of service classand has a requirement for bandwidth which can be actually provided inthe communication link by the communications network.

[0192] If such a codec has been determined, the codec, that is to saythe parameters which describe the transfer characteristic of the codec,is stored and the transmission parameters are supplied to a unit of thetransport layer which code data in accordance with the TCP.

[0193] If, however, it is not possible to transmit voice data via thecommunication network, a check is made whether the quality of servicevariable has the value 5, that is to say whether voice data are to betransmitted at all.

[0194] If this is so, the transmission parameters are stored as new,current values in accordance with the combined quality of service class5.

[0195] If this is not so, the values which are allocated to the combinedquality of service class 6 are stored as new transmission parameters.

[0196] In summary, the cycle of the method according to this exemplaryembodiment can be subdivided into three phases which will be graphicallydescribed in the text which follows.

[0197] In a first phase, every available codec is checked whether it canbe used at all for transmitting the desired data on the basis of thecurrently available bandwidth.

[0198] If this is so, an optimum LFP (Local Frames per Packet) parametervalue is determined for this codec as a function of the conflictingparameters delay and available bandwidth. In this first phase, a set ofsuitable optimally parameterized codecs is thus determined for thetransmission of the first data, in the form of pairs (codec, LFP).

[0199] In a second phase, each pair (codec, LFP) determined as suitablein the first phase is checked to see what quality of second data itpermits at a maximum. The quality of second data is quantified in theform of so-called multimedia conversion parameters.

[0200] Thus, the degrees of quality of chromaticity, granularity andimage size are determined which are appropriate for a given pair (codec,LFP) according to the current situations (available bandwidth, userprofile, negotiated session parameters etc.) and can be made available.

[0201] These values are combined in an MMC vector. As a result of thesecond phase, a set of suitable triples (codec, LFP, mmc_vector) is thusobtained.

[0202] The vector mmc_vector designates a vector of QoS parameters. Itrepresents an optimum point for the transmission in the space oftransmission quality classes for the second data spanned by thedimensions of chromaticity, granularity, image size etc.

[0203] In a third phase, the relatively best parameter set (code, LFP,mmc_vector) is then determined from the set of all triples determined inthe second phase by means of the current QoS measurement values and QoSpresettings.

[0204] The relatively best parameter set determined in this manner isconverted, i.e. used in the current transmission operation and the nextcycle can begin.

[0205] In the text which follows, the method for selecting the combinedquality of service class in the form of a pseudocode which is based onthe syntax of the programming language C++, is shown:

[0206] procedure AQuaVIT (String CONF_ID) /* CONF_ID is input parameter.CONF_ID contains an unambiguous designator for a particular conference(occasionally also called “session”). A conference involves two or moreend subscribers, at least one of which is mobile. Its terminal istypically a laptop, notebook or palmtop but, in principle, can also bean internet-capable mobile telephone. Other conference members areeither also mobile or communicate via a voice/data- enabled terminalwith a permanently networked Internet access. */ { //In next threeget-statements, the main input data for AQuaVIT are obtained.get(CONF_ID, Profile_Params); /* Fetching current profile parameters ofthe mobile end sub- scriber, among these priority factors for voice anddata streams, etc. */ get(CONF_ID, Stored_Params); /* Fetching storedQoS parameter values existing since the previous cycle, among these alsostored_NetworkType, etc. */ get(CONF_ID, AudioGwy_Params); /* Fetchingcurrent QoS parameters supplied by audio gateway, e.g. type ofmultimedia conversion (if dynamic), delay values, threshold values,packet loss rates, etc. */ boolean codecFound = false; // initializationof stop condition for white loop below int tryQOS; /* tryQoS is a QoSvalue according to column 1 in table 2 of the paper to be published.tryQoS is initialized with as high a value as possible on a trial basisin the next statement, as a function of the current profile parameters.This value will be gradually lowered later if, according to all othercurrent QoS parameters, there is no suitable codec which makes itpossible to have this QoS stage, until a suitable codec has been foundor until all available codecs have been unsuccessfully tried. It mustalso be noted that there can be a number of suitable codecs for each QoSstage and that each suitable one among the available ones is alsoidentified by AQuaVIT. At the end, the one whose actually the best onewill be selected among all suitable codecs which can guarantee thehighest possible QoS stage. */ /* Next, initialize tryQoS withconceivably highest value, depending on current QoS parameter in profileof mobile client */ initialize(tryQoS, Profile_Params); // tryQoS isinitialized, depending on QoS parameters in Profile /* The while loopbelow attempts to find, for a tryQoS value which is initially as high aspossible, a set of feasible codecs with correlated LFP (Local Frames perPacket) value. Each pair of a feasible codec and corresponding LFP valueis entered into a set. The loop iterates by degrading the value oftryQoS if no feasible codec has been found, and terminates if either afeasible codec has been found or none could be found at all. */ while(!codecFound) // codecFound is initialized by default to false { if(tryQoS ≧ 1.1&&tryQoS ≦ 4) // i.e., voice transmission is possible { */The for-loop below checks, for each codec with index i (1 ≦ i ≦numberOfavailableCodecs), if codec[i] is possibly feasible for tryQoS.If yes, then the most suitable LFP value corresponding to codec[i] iscomputed, and the pair (codec[i], LFP) is entered into the set offeasible pairs. Technically speaking, the QoS is the better the lessframes per packet are used. But the less frames are used, the morebandwidth is needed (headeroverhead). */ // first, initialize set offeasible pairs (codec, LFP) with the empty set set_of_feasible_pairs =empty_set; // in for-loop below, each feasible pair (codec[i], LFP) isentered into this set for (int i=0; i < numberOfavailableCodecs; i++) {/* Following if-cascade determines the optimal LFP parameter value forcodec[i]. Recall: The more frames there are per packet, the lessbandwidth is required but the higher the resulting delay is, i.e. thelower the resulting voice quality is. (Memo: in the commentary, thefollowing if-cascade is called “first pass”.)*/ ifpossibly_feasible(codec[i], Profile_params, Stored_Params,AudioGwy_Params) { bestLFP = compute_best_LFP(codec[i], Delay,Bandwidth); /* Compute best possible value of LFP for codec[i],depending on the Delay caused by LFP and the Bandwidth required by LFP.*/ set_of_feasible_pairs = set_of_feasible_pairs ∪ {(codec[i],bestLFP)}; // store the pair (codec[i], bestLFP) in set of feasiblepairs } if !is_empty(set_of_feasible_pairs) // negation (!) of is_emptyreturns boolean value { codecFound = true; } } //end for (int = 1; i <numberOfavailableCodecs; i++) if (!codecFound) // memo: this is stillthe ‘voice allowed’ case, i.e. tryQoS > = 1.1 && tryQoS < = 4 {decrement(tryQoS); // decrementation of QoS value is as described inpaper: // 1.1 => 1.2 => 1.3 => 1.4 => 3 => 5 and 2.1 => 2.2 => 2.3 =>2.4 => 4 => 6 } else // if codecFound = true /* Memo: This else blockcontains the phases called “second pass” and “third pass” in the abovecommentary. { set_of_feasible_triples = empty_set // initialisation //*for each feasible pair (codec[i], bestLFP), compute a feasible vector ofparameters for multimedia conver- sion of non-real-time data withrespect to estimated remaining bandwidth needed for non-real-time datatransmission, and add this vector as a triple (codec[i], best LFP,vector) to a set of feasible parameter triples. Memo: The followingfor-loop implements the “second pass” */ for each feasible pair(codec[i], LFP) in set_of_feasible pairs { mmc_vector =choose_feasible_MMC_parameters (remaining_bandwidth);set_of_feasible_triples = set_of_feasible_triples ∪ {mmc_vector}; } //Memo: The following statement stands for the “third pass”.bestCombination = choose_best_triple (set_of_feasible_triples); //choose best combination, i.e., the best of feasibles triples (codec,LFP, mmc_vector) store_new_QoS_values(CONF_ID, bestCombination); /*Previous statement stores best combination as new QoS parameter valuesfor CONF_ID } }// end if (tryQOS ≧ 1.1 && tryQOS ≦ 4), i.e., voicetransmission was allowed else // if tryQOS > 4; note that, in this case,no codec has been found! { if (tryQoS = 5 // i.e., no voicetransmission, and therefore no codec needed {store_new_QoS_values(CONF_ID, 5, Profile_Params, AudioGwy_Params); /*Previous statement computes and stores new QoS parameter values forCONF_ID according to QoS = 5 and according to Profile and currentAudioGateway parameters. */ } else //tryQOS = 6, i.e., no voicetransmission possible, no data transmission needed. {store_new_QoS_values(CONF_ID, 6, Profile_Params, AudioGwy_Params); }break; // while !codecFound would never stop unless broken in this elsecase, tryQOS > 4 . . . } }// end while !codecFound } // end AQuaVIT

[0207] The following documents were cited in the description above.Additional detailed information with regard to the foregoing descriptionmay be found in these publications:

[0208] [1] A. S. Tanenbaum, Computer-Netzwerke [Computer Networks],Wolframs's Fachverlag, 2^(nd) Edition, ISBN 3-925328-79-3, p. 17-32,1992

[0209] [2] IETF working group, PSTN and Internet Internetworking (pint),available in the Internet on Apr. 2, 2000 at URL address: http://www.ietf.org/html.charters/pint-charter.html

[0210] [3] M. Krautgärtner, H. Decker, et al., Design of V/D-API andArchitecture of the VE-MASE, CEC Deliverable NumberAC343/Siemens/WP2/DS/P/02/a1, Project Number AC343, November 1998

[0211] [4] ETSI TIPHON, Telecommunications and Internet ProtocolHarmonization Over Networks, General Aspects of Quality of Service(QoS), TR 101 329 V2.1.1 (1999-06), June 1999

[0212] [5] S. N. Bhatti, G. Knight “Enabling QoS Adaptations forInternet Applications”, Journal of Computer Networks, Vol. 31, No. 7, p.669-692, March 1999

[0213] [6] H. Decker, M. Krautgärtner, C. Ong, M. Wallbaum, Quality ofService Management in an Integrated Mobile Voice/Data-Enabled ServiceArchitecture, Proceedings of the 4^(th) ACTS Mobile CommunicationSummit, Jun. 8-11, 1999, Sorrento, Italy

[0214] [7] M. Handley et al., SIP: Session Initiation Protocol, IETFRequest for Comments 2543, March 1999

[0215] [8] WAP: Wireless Telephony Application Specification, availablein the Internet on Apr 2, 2000 at URL address:http://ww1.wapforum.org/tech/documents/SPEC-WTA-19991108.pdf

[0216] [9] H. Decker, M. Krautgärtner, Flexible Quality-of-ServiceTechnology for Supporting Voice/Data-Integrated Nomadic Networking, inB. Chistensen-Dalsgaard, W. Donelly, M. Griffith (eds), FlexibleWorking—New Network Technologies, IOS Press/Ohmsha, 1999, ISBN 1 58603028 0 (IOS Press), ISBN 4 274 90322 2 C3050 (ohmsha), Library ofCongress Card Number 99-67675

We claim:
 1. A method of transmitting data with real-time requirementand data without real-time requirement, which comprises: providing aplurality of first quality of service classes in an application layerfor transmitting first data with real-time requirement and a pluralityof second quality of service classes in the application layer fortransmitting second data without real-time requirement; selecting acombined quality of service class formed from the first quality ofservice classes and the second quality of service classes in theapplication layer, each combined quality of service class beingallocated transmission parameters specifying a transmission of the firstdata and the second data; and supplying the first data and the seconddata and the transmission parameters of the selected combined quality ofservice class to a unit of a transport layer, and transmitting the firstdata and the second data with the unit taking into consideration thetransmission parameters.
 2. The method according to claim 1 , whereinthe first data with real-time requirement contain voice data.
 3. Themethod according to claim 1 , wherein the second data contain dataselected from the group consisting of text data, video data, and imagedata.
 4. The method according to claim 1 , which comprises allocating toeach of the first quality of service classes a first priority and toeach of the second quality of service classes a second priority, andspecifying, based on the first and second priorities, a priority withwhich the first data and the second data, respectively, are to betransmitted.
 5. The method according to claim 4 , which comprisesforming the combined quality of service classes in dependence on thefirst and second priorities.
 6. The method according to claim 1 , whichcomprises selecting the combined quality of service class with thefollowing steps: a) selecting a combined quality of service class havingthe first quality of service class with a highest first priority and thesecond quality of service class with a highest second priority; b)checking whether a coder to be used can transmit the first data and thesecond data according to the transmission parameters of the respectivecombined quality of service class; c) if the checking step results in anaffirmative answer, selecting the combined quality of service class; d)if the checking step does not result in an affirmative answer, selectinga further combined quality of service class such that in each case thecombined quality of service class with reduced second priority isselected; and e) iteratively performing steps b) and d) until the codercan transmit the first data and the second data in accordance withtransmission parameters of the respective combined quality of serviceclass.
 7. The method according to claim 1 , which comprises coding andtransmitting the first data and the second data as a data stream with apredeterminable transport layer quality of service class in the unit ofthe transport layer.
 8. A communication device for transmitting firstdata with real-time requirement and second data without real-timerequirement, wherein a plurality of first quality of service classes areprovided in an application layer for transmitting the first data and aplurality of second quality of service classes are provided in theapplication layer for transmitting the second data, the devicecomprising: a processor programmed to select a combined quality ofservice class formed from the first quality of service classes and thesecond quality of service classes in the application layer, eachcombined quality of service class being allocated transmissionparameters specifying a transmission of the first data and the seconddata; and a transmission unit of a transport layer receiving from saidprocessor the first data and the second data and the transmissionparameters of the selected combined quality of service class, andtransmitting the first data and the second data taking intoconsideration the transmission parameters.
 9. The communication deviceaccording to claim 8 configured as a mobile communication device.
 10. Acommunications system, comprising said communication device according toclaim 8 configured as a first, mobile communication device, and a secondcommunication device, wherein the first data and the second data can betransmitted from said first communication device to said secondcommunication device.